One of the most popular, and network intensive, computer applications is video or multimedia playback over the internet. Several years ago, it was believed that the internet, with its aging protocols and lack of guaranteed delivery, would be unable to support video playback at a quality satisfactory to users. However, applications such as YouTube and Netflix have shown that quality video playback is possible.
There were various mechanisms that were created to improve the user experience. For example, in some embodiments, the client, which has a media player, requests and receives, the entire multimedia file before beginning playback of the file. This insures that, once the file begins playing, it will be able to continue uninterrupted. While this may be acceptable for shorter clips, such a scheme is unacceptable for longer files for several reasons. First, the user is forced to wait until the entire file is downloaded before seeing any portion of the clip. Such a wait may be unacceptable. Second, in an environment with limited network bandwidth, the entire file is transmitted, even if the user only watches the first few seconds of the video.
Another method is known as progressive download. In this embodiment, the client begins downloading the multimedia file from the server. When a certain threshold is reached, such as 3 seconds of the download has been completed, the media player on the client begins displaying the video. The threshold used may be fixed, or may be based on the resolution of the file, the average available network bandwidth or other parameters. This has several advantages over the previous method. First, the user does not need to wait until the entire file is downloaded before beginning to view the video. Secondly, bandwidth is potentially saved if the user chooses to navigate away from the video before it is completely downloaded. However, one shortcoming of this mechanism is that once the video has begun playing, the available bandwidth must remain above a minimum level to insure that the downloading of each subsequent portion of the video completes before that portion needs to be displayed. If the bandwidth decreases, the video may appear choppy, or may pause in order to allow the buffer to fill again.
Progressive download methods typically download the multimedia content at the maximum rate supported by the underlying transport within the limits of the player buffer (i.e. limiting the downloaded content not to overflow the player buffer by TCP acknowledgement mechanisms). These methods improve the quality of experience (QOE) by reducing stutters since the content is transferred to the client buffer, thus minimizing the possibility of player stopping during content presentation. However, one disadvantage is that when the transit network bandwidth is high compared to the media stream rate, significant content is downloaded to the player, and if the user cancels the current presentation, such as by moving on to a different media-clip, the network bandwidth used for the downloaded content that is not viewed is wasted.
Another mechanism used by Adobe Systems, Inc. is known as Real Time Messaging Protocol (RTMP). This mechanism controls content delivery using RTMP protocol over TCP, by controlled streaming of the content. Thus, the streaming mechanism maintains a window of what is delivered to the player relative to what is currently being played to the user. Thus bandwidth wasted due to user cancellations of active sessions is reduced, by limiting the content delivered to locations relatively close to the current display position. RTMP has also options to use HTTP or HTTPS as transport options.
Another mechanism is Real Time Streaming Protocol/Real-time Transport Protocol (RTSP/RTP). In this protocol, the server controls the rate of delivery of the content to the presentation rate, rather than delivering at the maximum rate of the underlying transport, such as TCP or UDP over IP.
In addition, several HTTP streaming protocols have been defined to deliver live or stored multimedia content at controlled rate (determined by the server or based on cooperation between the client and server) that matches the multimedia stream rate.
RTMP and HTTP streaming methods pace the content delivery, by limiting the rate at which content is delivered. RTMP uses protocol components in the client and the server, where as HTTP Streaming uses additional tags in the HTTP Requests and Responses.
In Adaptive Bit Rate Streaming (ABR Streaming), the client monitors the bandwidth to the server at the start up of multi-media session and at regular intervals during the multi-media play. Based on the monitored bandwidth, the client selects alternative encodings with different screen resolutions of the same content. The switching between alternative resolutions of the same media content is done explicitly by the client. Several applications, such as Ankeena TV, use ABR streaming over HTTP for live content delivery.
However, in this protocol, it is the client that determines the available bandwidth. Thus, in a radio access network, where devices are routinely added and removed from a particular cell, the client's limited knowledge of the overall network utilization may compromise its ability to accurately predict a suitable encoding and screen resolution in a timely manner.
FIG. 1 shows a traditional 3G/UMTS network, including an UE (user equipment) 107, NodeB (or base transceiver station) 106, RNC (radio network controller or base station controller) 105, SGSN (Serving GPRS support node) 104, and a GGSN (gateway GPRS Support node) 103. Also shown in FIG. 1 is the protocol used to communicate between these various devices. For example, IuB 108 is the protocol used between Node B 106 and the RNC 105. Similarly, IuPS is the protocol used between the RNC 105 and the SGSN 104. Gn 110 is used between the SGSN 104 and the GGSN 103. Finally, Gi 111 is the IP based interface between the GGSN 103 and the internet.
It would be advantageous if a component in the radio access network, with visibility to total network traffic, were able to determine when a particular user was requesting multimedia content and based on that total network traffic, were able to configure the encoding and resolution of that multimedia content to maximize the user's quality of experience. In addition, it would be beneficial if that component could decouple wireless RAN traffic from core network traffic, by buffering multimedia content and delivering this content in a just-in-time manner to the end user device.